Asterisk - The Future Of Telephony

What is Asterisk?

Asterisk is an open source PBX, or "Private Branch eXchange", but such a simple description barely does it justice. Asterisk is, in fact, the leading telephony engine and tool kit in the world - released under GPL or "General Public License", and therefore available for download free of charge - allowing flexible communications solutions be created by developers and integrators alike. Not only is the Asterisk software itself, free, but it runs on Linux BSD ("Berkeley Software Distribution", a.k.a. "Berkeley Unix") emulated Microsoft Windows and Macintosh OS X such that it can offer interoperability with almost all standards-based telephony equipment at relatively low cost.

How and Where is Asterisk Deployed?

Asterisk can be deployed as a gateway, as a feature or media server, or in a call centre where it can provide the backbone of a complete ACD or "Automated Call Distribution" system for example. Asterisk can provide a bridge between the existing PSTN ("Public Switched Telephone Network") and IP or VoIP (Voice over Internet Protocol") telephony which is rapidly becoming a mainstream service. Asterisk supports a wide range of telecommunications standards and protocols - H.323, SIP ("Session Initiation Protocol"), MGCP ("Media Gateway Control Protocol"), etc. - and its modular design means that it can convert from one to another with relative ease. In fact, a total of four APIs ("Applications Programming Interfaces") are defined as loadable modules so the Asterisk core itself, does not deal with call connection, codec translation and other functions that can be performed elsewhere.In terms of features and functionality, Asterisk is streets ahead of many proprietary systems including those at the high end of the market. Conference bridging, IVR ("Interactive Voice Response"), auto attendant and voicemail capabilities are all included, as is Unified Messaging which allows voice, fax, SMS and email messages to be accessed via a single mailbox.

How is Asterisk Used by Telecoms Service Providers?

Telephone service providers can take advantage of Asterisk for the provision of feature servers, voicemail systems, prepaid calling solutions, etc., all of which are more flexible and less expensive than those provided by alternative means. Asterisk is deployed on millions of servers worldwide to manage VoIP telephony for consumers and businesses, for example and requires no additional hardware. In fact many VoIP service providers nowadays, not only support Asterisk, but are explicitly designing their own services to work with Asterisk. Incoming and outgoing calls can be handled by different VoIP or telecoms service providers if need be, which can be useful if one telephone service provider allows only incoming VoIP calls to avoid the provision of directory services, etc.. It may be of course that a telephone service provider boycotts VoIP altogether, but this need not necessarily preclude the use of VoIP; Asterisk can convert calls between TDM ("Time Division Multiplexing") and VoIP, and back again, as they leave or enter the carrier network. Asterisk also supports a range of hardware for the connection of existing digital and analogue telephony equipment.

Source : www.articlesbase.com

Open the door to telephony's future - Technology Information

Communications News , Sept, 2000 by Bryan Pelham

Innovation depends on standards promoting Internet voice services.
A revolution is taking place today in telephony that mirrors the Internet revolution of the past decade. Slowly, but surely, carriers are building out a converged voice/data infrastructure that will foster an explosion in new telephony services and applications. To achieve full voice/data convergence, open standards will be needed, standards that are supported by everyone in the voice/data food chain, from carriers and service providers to businesses and consumers. These standards are emerging quickly, and now is the time to plan for them.
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To foster innovation in telephony networks, all parties must work together to transform the public switched telephone network (PSTN) from a closed, proprietary system optimized for voice delivery into a converged voice/data network based on open standards. Today, PSTN delivery platforms, such as Class 5 switches, are built on proprietary architectures, and years are needed to deploy new features on these architectures.
In addition, carriers deploy PSTN switches from various vendors, each with its own proprietary way of implementing features--impeding the smooth, uniform rollout of services. For example, caller ID, three-way calling and call return have been technically possible for many years, but these and other features have been painfully slow to appear, due to lack of standardization. As a result, carriers have difficulty performing across-the-board software and feature upgrades needed to offer a service universally.
In contrast, the use of Internet standards for the deployment of voice services will enable the same kind of rapid innovation for telephony that is occurring on the Internet today. Just 10 years ago, few people had even heard of the Internet, but today those same people wonder how they could live without it. The Internet touches lives on a daily basis--either directly as e-mail is sent or received, as company Web sites are explored, or as people check their investment portfolios; or indirectly when goods and services are purchased from companies that rely on Internet standards to conduct business.
In the telecommunications industry, the following Internet standards are taking root:
* The media gateway control protocol (MGCP) and the newer media gateway control protocol (MEGACO) are used for controlling voice over Internet protocol (VoIP) gateways from external call control agents. These protocols assume a call-control architecture where the call-control "intelligence" is outside the gateways and handled by external call-control agents.
* The session initialization protocol (SIP) is an application-layer control (signaling) protocol used between intelligent devices for creating, modifying and terminating sessions with one or more participants. SIP is designed to be independent of the lower-layer transport protocol and can be extended with additional capabilities.
* Although not an Internet standard, the H.323 protocol has also been widely deployed to build VoIP networks. H.323 is also an application-layer control protocol used between intelligent devices for managing sessions with other multimedia end points.
Voice and data networks will converge around the IP standard, as well as a host of other Internet standards for multimedia applications, such as IPtelephony and streaming video. Incumbent and competitive carriers alike are aggressively moving to IP and softswitch technologies that will enable them to implement new features quickly through a single, cost-effective network.
With the carriers moving toward convergence, network managers should also begin planning for voice/data network integration around IP. Information analysts at Phillips Group-Info Tech estimate that nearly 90% of enterprises with multiple locations will begin migrating from the circuit-switched public switched telephone network (PSTN) to packet-based telephony over the next five years.
THE PROMISE OF CONVERGENCE
The benefits of voice/data convergence accrue across the board, from the carrier network to the customer premises:
* Lower capital equipment costs. Companies and carriers will lower their infrastructure and management costs with a single, integrated voice/data network infrastructure. Telephone and data equipment is increasingly becoming integrated. For example, many companies are replacing multiple voice and data devices with integrated communications platforms (ICPs) that provide a multitude of services in one box.
* Lower maintenance costs. By merging voice and data networks, carriers and businesses reduce management requirements and personnel needs by supporting only one converged network, instead of separate voice and data networks. Business customers also reduce administrative overhead with only one provider relationship to maintain.
* Better bandwidth usage. Circuit-switched voice networks frequently waste available bandwidth because voice lines sit idle for a significant portion of each day. By using a packet-based network for telephony transport, carriers and companies can reduce costs by leveraging unused and expensive wide area network bandwidth for data services. For example, idle voice bandwidth can be used for large overnight data transfers or backups.

Source : http://findarticles.com

What is computer Telephony Software?

Computer telephony software can be divided into a number of categories, each of which provides a set of functionality for Computer Telephony Integration (CTI).


Automatic Call Distributors (ACD's)
Automatic Call Distributors are used by inbound call-centers to evenly distribute telephone calls between available customer service representatives.
Interactive Voice Response (IVR)
Interactive Voice Response systems are used to gather data from callers before transferring them to a customer service representative.
IVR systems are well known for their frequent repetition of "Press or say 1 for ...".
Screen Population
Screen pop programs take Caller-ID or ANI data and information from IVR systems and display it on the computer screen.
Software Dialers
Software dialers enable users to dial the telephone directly from their computer. This is very convenient, because computers can store address books of thousands of contacts.
Autodialers
Autodialers are a category of software dialers that are designed to dial a large volume of telephone numbers very quickly.
Predictive Dialers
Predictive dialers are a second generation of autodialer technology.
Predictive dialers examine the availability of a group of outbound call center agents and determine the number of telephone calls to place based upon the number of available agents.
Predictive dialers reduce the time that telemarketers spend waiting for calls or talking to answering machines or fax machines.
Sometimes, the predictive dialer will dial more telephone calls than there are available call agents. If you have ever received a call from a telemarketer and been asked to wait to speak with an agent, you have experienced this type of predictive dialer failure.
Computer Answering Machine Software
Computer answering machine software enables a computer to act as an answering machine. Custom greetings can be defined, IVR technologies can be integrated, and voice messages can be saved to disk, e-mailed, or sent across a network for remote access.
Voice Mail Boxes
Voice Mail Boxes act as computer answering machines for large organizations. A VMB may support thousands of individual voice mail boxes.
PBX's
Computer-based PBX systems enable even the smallest companies to maintain their own Private Branch Exchange switches at costs significantly lower than traditional dedicated hardware PBX systems.

Source : www.tech-faq.com

Signaling System 7

Introduction
SS7 is a critical component of modern telecommunications systems. SS7 is a communications protocol that provides signaling and control for various network services and capabilities. While the Internet, wireless data, and related technology have captured the attention of millions, many forget or don't realize the importance of SS7. Every call in every network is dependent on SS7. Likewise, every mobile phone user is dependent on SS7 to allow inter-network roaming. SS7 is also the "glue" that sticks together circuit switched (traditional) networks with Internet protocol based networks.
SS7 Technology
SS7 signaling is a form of packet switching. Unlike circuit switching, which utilizes dedicated data "pipes" for transmission of information, packet switching dynamically assigns "routes" based on availability and "least cost" algorithms. Another example of packet switching is TCP/IP, the protocol used for routing messages over the Internet. Unlike the Internet, which utilizes a vast public "web" of interconnecting facilities and routing equipment, SS7 networks are private and logically self-contained. The private nature of SS7 networks is critical for security and reliability.
SS7 involves two different types of signaling: connection oriented signaling and connectionless oriented signaling. Connection oriented signaling refers to the establishment of switch-to-switch facilities call inter-office trunks. These trunks carrier voice communications. The ISDN User Part (ISUP) part of the SS7 protocol is utilized to establish trunks between switches. In contrast, the Transaction Capability Application Part (TCAP) is utilized for connectionless signaling which typically entails switch-to-database or database-to-database communications. An example of connectionless signaling is TCAP signaling of HLR to VLR communications discussed in the mobile networking article.
SS7 Networks
SS7 is comprised of a series of interconnected network elements such as switches, databases, and routing nodes. Each of these elements is interconnected with links, each of which has a specific purpose. The routing nodes are the heart of the SS7 network and are called a Signal Transfer Point (STP). STPs are connected to Service Switching Points (SSP) that are switches equipped with SS7 control logic. SSP switches are connected to the STPs via Access links (A links). STPs also connect to databases called Service Control Points (SCP) via A links. The SCP is the network element that contains service control logic such as instructions for converting a 8XX (toll-free) number into a routable number.
STPs are always deployed in pairs, allowing a spare should one of the STPs have a problem. Each STP of a "mated pair" are connected to each other via Cross links (C links). STP pairs connect to other STP pairs via Bridge or Diagonal links (B or D links). B links connect STP pairs that are at the same level of hierarchy while D links connect STP pairs that are different hierarchial levels. An example would be STPs in a local network connecting with STPs of a long distance network. Being at different hierarchies, the local-to-long distance links would be considered D links.
Links used for SS7 communication directly between SSPs (no STP involved) are called Fully associated links (F links). An example of these links are those that are used in combination with voice trunks between two mobile network SSPs. The F link is used to signal a hand-off message from one SSP to the other, allowing the mobile phone user to travel from one area (served by one switch) to another area (served by another switch).
Extended links (E links) are used to connect an SSP to an alternative STP pair. In the event that the primary STP pair is inoperable, the alternative pair establishes operations with the SSP over the E links.
Business Issues
In today's modern telecommunications networks, SS7 is used for virtually every call to establish a voice connection between the calling and called party locations. SS7 is also the medium for advanced capabilities and applications including mobile networking and services as well as wireline applications such as toll-free calling and automatic calling card identification.

Source : www.mobilein.com